[asterisk-users] scaling with SMP
Is there a way to cause asterisk to benefit from running on a machine with more than two cores? I only see two processes running, with one at a very low priority and the other at a very high priority. I'm guessing one is managing the other. Thanks, Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] disabling authentication
Is there a way to cause asterisk to accept all calls without any authentication? Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk startup is slow
Hi, Having moved from asterisk-1.2.8 to 1.2.14, i've noticed that startup is much slower. In other words, if I say asterisk -R they type stop now, it takes on the order of 7 seconds instead of 1 second. The old asterisk startup printed out something like 650 lines, whereas the new one prints out lsomething like 720 lines. Have other people been seeing this? Is there something I have done wrong? As far as I know, both were built with make all make install (i.e. no compile time options being set differently). tia Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] proxy howto
Hi, I've been trying to get asterisk to use an outbound sip proxy. Putting the outboundproxyhost directive in the [general] section of sip.conf, but it doesn't seem to work. My expectation would be that by setting outboundproxy and outboundproxyport in that location, then all dial commans (or at least dial commands of the form Dial(SIP/[EMAIL PROTECTED]) or Dial(SIP/[EMAIL PROTECTED])) would automatically be forwarded to the proxy. However, this is not happening. In other words, the INVITE packet generated by a Dial() are going straight to the destination domain. Similar results for putting it in a [peer] section. Thanks Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: sip address in voicemail emails
Hi, Anselm On 12/2/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Freitag, den 01.12.2006, 17:57 -0500 schrieb Mark Price: Hi, On 12/1/06, Mark Price [EMAIL PROTECTED] wrote: hi, I am using asterisk 1.2.10. I am trying to send sip links in asterisk voicemail, so that users can easily reply to emails. This does not seem to be straightforward. First, there seems to be no variable that prints out the domain name of the sip call, since I am including every variable mentioned on http://www.voip-info.org/wiki-Asterisk +config+voicemail.conf You might use your own notification script, s.t. you can send a Content-type: text/html for your E-Mail. I don't understand how the notification script is useful for this purpose. The voicemail.conf page on voip-info.org that is referenced above says the following: The way it works is basically any time that somebody leaves a voicemail on the system (regardless of mailbox number), the command specified for externnotify is run with the arguments (in this order): context, extension, and number of voicemails in that mailbox In other words, the the documentation says that the externnotify command is not given any information at all regarding the source of the phone call. Thanks, Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip address in voicemail emails
hi, I am using asterisk 1.2.10. I am trying to send sip links in asterisk voicemail, so that users can easily reply to emails. This does not seem to be straightforward. First, there seems to be no variable that prints out the domain name of the sip call, since I am including every variable mentioned on http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf Second, if i include in the emailbody variable sip:[EMAIL PROTECTED] (on the wrong assumption that the phone call was from a telephone number), gmail responds by assuming it is a mailto link, and outlook treats it as plain text. Having examined the emails being sent in mutt, it appears that the message has no mime type (neither text nor text/html). Unsurprisingly, then, enclosing it in an href, i.e. a href=sip:[EMAIL PROTECTED]click herea/ is of no help. Is there a workaround for this? Perhaps it is being addressed in a later version of asterisk? I have seen no mention of it in the release notes for 1.4. Thanks Mark Price ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: sip address in voicemail emails
Hi, Item zero, Thanks for hosting and participating in this list, digium, and all the developers involved. First, I realize that my first post probably did not belong on asterisk-dev. I had intended to remove that address from my recipients list, but did not Secondly, I figure that now that my neck is extended, I should write to clarify one thing and correct another in my first email. On 12/1/06, Mark Price [EMAIL PROTECTED] wrote: hi, I am using asterisk 1.2.10. I am trying to send sip links in asterisk voicemail, so that users can easily reply to emails. This does not seem to be straightforward. First, there seems to be no variable that prints out the domain name of the sip call, since I am including every variable mentioned on http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf To clarify to myself, I made a call from a different sip domain from a username that does not exist on the asterisk box, and found out that it is true: VM_CIDNUM contains the username, but not the domain name of the call. Therefore, as long as the username is a telephone number, we can work around that, but the message printed to describe a non-telephone-number phone call will be incorrect. Second, if i include in the emailbody variable sip:[EMAIL PROTECTED] (on the wrong assumption that the phone call was from a telephone number), gmail responds by assuming it is a mailto link, and outlook treats it as plain text. Having examined the emails being sent in mutt, it appears that the message has no mime type (neither text nor text/html). I thought I should point out I was incorrect. The text portion of the email is given mime-type text. Therefore it appears to be impossible to send a sip link in the email unless the receiving email client knows how to recognize them (as many know how to recognize http:// and mailto: links). Thanks, Mark Price ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] specify codec by domain?
Hi, My pstn provider is currently set up so that when asterisk sends an outbound SIP call to them, if sip.conf says: [general] allow=g729,ulaw then it always picks ulaw, even though g729 is listed first. However, if sip.conf says: [general] allow=g729 then g729 is chosen. Dial([EMAIL PROTECTED] How can I force calls to this sip domain to use g729? Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] autocreate peer + sippeers table entry = auth required?
In my setup, sip calls coming in through a proxy with a sip.conf entry set to autocreatepeer=yes and context=proxy get placed into context proxy in the dial plan. That is expected.However, if the username in the From: address exists in the sippeers table, it gets challenged for the password and on success is dropped into context default, even if the sip domain is not being served by asterisk. What could be the problem?Here is my sip.conf:[general]Autocreatepeer = nocontext=defaultdomain = b.comdomain = sip.b.com realm=b.combindport=5060bindaddr=4.2.2.2allow=g729,ulaw,alaw,speex,gsmdtmfmode=rfc2833rtcachefriends=yes;bindaddr=0.0.0.0 srvlookup=yesrtpkeepalive=1000rtupdate=yesport=5060defaultexpirey=3600tos=0x18insecure=no[ser]type=peercontext=serhost=4.2.2.3canreinvite=yes inseure=veryautocreatepeer=yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] external username conflict in dialplan
I'm seeing an interesting problem in asterisk:asterisk has domain a.com and the sip proxy has domain b.com.The sip proxy is configured as a friend in sip.conf.
If a call comes in to asterisk from the sip proxy, if ${EXTEN} exists in the sippeers table the call goes to the default context else the call goes into the ser contextWhy would that happen? Is this expected?
sip.conf has:[ser]type=friendcontext=proxyhost=66.129.95.23canreinvite=yesautocreatepeer=yes;monitored=truecontext=serThanksMark
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Re: [asterisk-users] sequential Dial() commands
I was using timeouts. The dial plan was altered and working by the time of the replies, so I'm sorry I can't show the original problem.It's possible I did something simple like not waiting the whole timeout.Thanks, MarkOn 10/10/06, Dovid B [EMAIL PROTECTED] wrote: Simple Exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],90) ; This will ring thier phone for 90 seconds Exten =1234,2,Noop(If user dosent pick up do something here) Exten = 1234,102,Dial(SIP/[EMAIL PROTECTED],90) ; WIll ring user B if User is Busy or hits the reject button Exten = 1234,103,Noop(If user dosent pick up do something here) Exten = 1234,203,Noop(If user B is busy or rejects call do something here) - Original Message - From: Mark Price To: Asterisk Users Sent: Tuesday, October 10, 2006 7:28 PM Subject: [asterisk-users] sequential Dial() commands Hi,How do I cause the dial plan to dial a different extension if the first either never picks up or presses ignore or what have you?For example, something like this:exten = context,1,Dial( SIP/[EMAIL PROTECTED])exten = context,2,Dial( SIP/[EMAIL PROTECTED])Currently, if the first number doesn't answer, the session is closed.ThanksMark ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sequential Dial() commands
Hi,How do I cause the dial plan to dial a different extension if the first either never picks up or presses ignore or what have you?For example, something like this:exten = context,1,Dial( SIP/[EMAIL PROTECTED])exten = context,2,Dial(SIP/[EMAIL PROTECTED])Currently, if the first number doesn't answer, the session is closed.ThanksMark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] authenticating forwarded calls
Is it possible to set up SER with asterisk so that any INVITE that is forwarded from SER to asterisk does not need to be authenticated by asterisk? Thanks, Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk queues with SER, aka sip show peers
Hi,I am trying to integrate asterisk queues with SER.We have our queues set up in the following manner:An entry in the queue members table consts of the queue name and a SIP address.For example,queuename | member -support-q | SIP/5558675309We have observed the following behavior:If the first phone is registered directly to asterisk, then the queue knows about the phone, and puts it in the queue rotation. I would like to get this same functionality using SER as the registration server.Currently, if a phone is registered with the SER proxy, * knows nothing about it until the phone calls an extension on the asterisk server. After that, the phone shows up in sip show peers, but with address (unknown).In neither case is the queue aware of the phone.Here is the setup:Asterisk is set up in general to be real time, including sip peers and sip users. in sip.conf, [general]... rtcachefriends=yes register = [EMAIL PROTECTED]: [EMAIL PROTECTED]/2342342345[ser]type=friend insecure=veryhost=sip.proxy.comcanreinvite=yesautocreatepeer=yescontext=proxyAsterisk is successfully registering with SER, and sip show peers shows the SER box as a peer, although I think that is irrelevant of if the SER service is running or not. What do I need to do to get Asterisk to be aware of the clients?Mark Price ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple authentication realms
Hi, I'm a little bit confused by the documentation regarding multiple-realm authentication. Asterisk-1.2.9.1/CHANGES says: Since 1.2.0-beta1: snip * SIP domain support for authentication and virtual hosting However, the documentation in sip.conf seems to suggest that only one realm may be specified. The confusion lies in the fact that virtual hosting usually implies the ability to host multiple domains within one server. Is it possible to have multiple authentication realms as the document suggests, or am I misunderstanding? Thanks, Mark Price ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dial plan return values
Is there a method for detecting return values of applications in the dial plan? Thanks Mark Price UNETA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
